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WO1992017942A1 - Procede de codage de signaux numeriques - Google Patents

Procede de codage de signaux numeriques Download PDF

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Publication number
WO1992017942A1
WO1992017942A1 PCT/JP1992/000237 JP9200237W WO9217942A1 WO 1992017942 A1 WO1992017942 A1 WO 1992017942A1 JP 9200237 W JP9200237 W JP 9200237W WO 9217942 A1 WO9217942 A1 WO 9217942A1
Authority
WO
WIPO (PCT)
Prior art keywords
band
critical band
critical
block
bits
Prior art date
Application number
PCT/JP1992/000237
Other languages
English (en)
Japanese (ja)
Inventor
Kenzo Akagiri
Original Assignee
Sony Corporation
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Priority claimed from JP3091188A external-priority patent/JP3060577B2/ja
Priority claimed from JP3091187A external-priority patent/JP3060576B2/ja
Application filed by Sony Corporation filed Critical Sony Corporation
Priority to RU92016428/09A priority Critical patent/RU2090973C1/ru
Priority to EP92906204A priority patent/EP0531538B1/fr
Priority to DK92906204T priority patent/DK0531538T3/da
Priority to BR9204799A priority patent/BR9204799A/pt
Priority to DE69225100T priority patent/DE69225100T2/de
Priority to AU13383/92A priority patent/AU654533B2/en
Publication of WO1992017942A1 publication Critical patent/WO1992017942A1/fr
Priority to NO92924552A priority patent/NO924552L/no
Priority to FI925400A priority patent/FI109627B/fi

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Classifications

    • H—ELECTRICITY
    • H03—ELECTRONIC CIRCUITRY
    • H03M—CODING; DECODING; CODE CONVERSION IN GENERAL
    • H03M7/00—Conversion of a code where information is represented by a given sequence or number of digits to a code where the same, similar or subset of information is represented by a different sequence or number of digits
    • H03M7/30—Compression; Expansion; Suppression of unnecessary data, e.g. redundancy reduction
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B1/00—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission
    • H04B1/66—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission
    • H04B1/665—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission using psychoacoustic properties of the ear, e.g. masking effect

Definitions

  • the present invention relates to a digital signal high-efficiency encoding method or apparatus for encoding an input digital signal.
  • input signals such as audio and audio are divided into signal components of multiple channels on the time axis or frequency axis, and each channel has its own signal.
  • bit allocation There is a coding technique based on bit allocation (bit allocation) that adaptively allocates the number of bits.
  • bit allocation bit allocation
  • the above-mentioned coding technology based on bit allocation of audio signals and the like includes band division coding (sub-band coding) in which an audio signal or the like on the time axis is divided into signal components of a plurality of frequency bands and encoded.
  • So-called adaptive transform coding (ABC) which transforms a signal on the time axis into a signal on the frequency axis (orthogonal transform), divides it into multiple frequency bands, and adaptively codes each band.
  • APC adaptive predictive coding
  • an audio signal on the time axis or the like is time-transformed at predetermined unit time intervals by an orthogonal transform such as a fast Fourier transform (FFT) or DCT. It is converted to an axis (frequency axis) orthogonal to the axis, then divided into a plurality of bands, and the coefficient data of each of these divided bands is encoded by adaptive bit allocation. This encoded data is transmitted.
  • an orthogonal transform such as a fast Fourier transform (FFT) or DCT.
  • the configuration for later decoding includes the coefficient data that has been subjected to the above-described band division and block-blocking processing for each block, and the floating coefficient and the number of allocated bits for each block.
  • the present invention has been proposed in view of the above situation, and has as its object to provide a digital signal encoding method capable of higher bit compression.
  • a digital signal encoding method has been proposed to achieve the above-mentioned object.
  • An input digital signal is orthogonally transformed and divided into critical bands, and the digital signal is set based on the energy of each critical band.
  • the signal component of each critical band is encoded with the number of bits according to the level of the difference between the allowable noise level of each critical band and the energy of each critical band, and the signal component after the orthogonal transformation is encoded.
  • This is a digital signal encoding method for performing block floating processing for each block, performing block floating processing for each block, and transmitting floating coefficients for each block.
  • the present invention When performing the block floating processing in units of small blocks in a band narrower than the critical band, information on the permissible noise level is transmitted instead of the floating coefficient for each critical band, and assigned to each small block. Word length information according to the number of bits Is transmitted. Further, when performing the block floating process in units of large blocks having a band wider than the critical band, the present invention transmits information on the floating coefficient of each large block and allocates the information to each of the critical bands. The information of the code length according to the number of bits to be transmitted is transmitted.
  • the block floating process when the block floating process is performed in small blocks in a band narrower than the critical band, the node length according to the number of bits allocated to one of the small blocks in each critical band is set. It is designed to transmit information.
  • the block floating process when the block floating process is performed in a large block having a band wider than the critical band, the code length information of one of the critical bands in the large block and the allowable noise level of each critical band are used. It is also possible to transmit information about and transmission.
  • information on an allowable noise level may be transmitted by shifting a specified range lower by a predetermined level from a signal level range in the critical band.
  • a so-called masking amount is determined from the energy of each critical band in consideration of human auditory characteristics, and each critical amount is determined using an allowable noise level based on the masking amount. It is desirable to determine the number of bits allocated to the band.
  • DISCLOSURE OF THE INVENTION when the block floating process is performed on a small block in a band narrower than the critical band, a plurality of small blocks exist in one critical band. Without transmitting the floating coefficient for each small block By transmitting the information on the allowable noise level and the word length information for each critical band, the number of bits for the floating coefficient can be reduced.
  • the other small blocks within this critical band can be transmitted. Therefore, the number of bits for transmitting the guide length information can be reduced.
  • FIG. 1A is a diagram for explaining a case where the floating process of the present invention is performed with a block narrower than a critical band.
  • FIG. 1B is a diagram for explaining a case where the floating process of the present invention is performed on a block wider than the critical band.
  • FIG. 2 is a flowchart of the embodiment of the present invention.
  • FIG. 3 is a block circuit diagram showing a specific configuration for setting an allowable noise level.
  • FIG. 4 is a diagram showing a bark spectrum.
  • FIG. 5 is a diagram showing a masking spectrum.
  • Figure 6 is a diagram that combines the minimum audible curve and the masking spectrum.
  • FIG. 7 is a diagram for explaining an allowable noise level transmitted by each block according to the present embodiment.
  • the digital signal encoding method performs a quadrature transform of an input digital signal by, for example, a fast Fourier transform (FFT) or a DCT, and a higher frequency band considering human hearing characteristics.
  • FFT fast Fourier transform
  • DCT digital to analog converter
  • critical bands B B1 to B4 that increase the width, and the allowable noise level NL for each critical band set based on the energy for each critical band and the critical band for each critical band
  • the signal components (coefficient data) of each of the critical bands are encoded with the number of bits corresponding to the level of the difference from the energy of each block, and the signal components after the orthogonal transform are converted into the signal components of each block, so that each block b
  • This is a digital signal encoding method in which block floating processing is performed in advance, and a floating coefficient Fc for each block b is transmitted.
  • step S1 the signal components after the orthogonal transformation are divided into blocks, and block floating processing is performed for each block to determine a floating coefficient (floating level) for each block.
  • step S2 the allowable noise level for each critical band is determined based on the energy for each critical band as described later.
  • step S3 the allowable noise level for each critical band and each critical band are determined.
  • the information on the code length is determined according to the number of allocated bits set based on the level of the difference from each energy. In this embodiment, for example, as shown in FIG.
  • FIG. 1A shows one extracted critical band B on the high band side where the bandwidth of the critical band is wide.
  • FIG. 1B shows the band on the low band side where the bandwidth is narrow.
  • Each band is shown.
  • FIGS. 1A and 1B since the level difference for obtaining the number of allocated bits and the code lengths W1 to W4 correspond to each other, the code lengths W1 to W4 are referred to for convenience. Is shown in the figure.
  • encoding processing of FFT coefficient data by adaptive bit allocation is performed. That is, the encoding process by adaptive bit allocation in the present embodiment obtains a so-called masking amount in consideration of human auditory characteristics as described later on the basis of the energy of each critical band.
  • the number of allocated bits is determined according to the level of the difference between the allowable noise level set based on the amount of noise (that is, the substantially constant allowable noise level for each critical band) and the energy of each critical band.
  • Encoding is performed in accordance with the number of allocated bits for each.
  • each of a plurality of coefficient Blocks are configured, and so-called block floating processing is performed for each block to perform bit compression. Therefore, as shown in the example of FIG.
  • a plurality of the blocks (for example, the four small blocks b1 to b4) exist in the high critical band, that is, the critical band B having a wide bandwidth.
  • the bandwidth is narrow (lower than the critical band) as in the example of Fig. 1B, multiple critical bands (for example, four critical bands B1 to B4) exist in one large block b. I will be.
  • the floating coefficient of the floating process and the word length information according to the above allocated bit number are required. Become. That is, for the configuration for later decoding, usually, the information of the floating coefficient for each block is used.
  • a system configuration for transmitting information of a code length corresponding to the number of allocated bits based on the level difference between the level of the floating coefficient and the allowable noise level of the critical band In other words, at the time of subsequent decoding, the most significant bit (MSB) in the block floating processing is determined from the information of the floating coefficient, and the least significant bit (LSB) is determined from the word length information. Thus, the allowable noise level is determined. Further, the coefficient data (main data) of each block and the magnitude of the signal are determined.
  • the information of the floating coefficient is represented by, for example, 6 bits
  • the information of the above-described code length is represented by, for example, 4 bits.
  • the orthogonal transform is DFOT (discrete Fourier transform)
  • the word length information indicates the magnitude (amplitude) and phase or the real part and the imaginary part by the four bits. Therefore, for example, when one critical band is divided by a plurality of floating blocks, the total number of transmission bits in the critical band according to the number of blocks of the block floating process (that is, the number of divided bands) is , As shown in Table 1.
  • the information of the floating coefficients Fc1 to Fc4 of the small blocks in the critical band B is not transmitted, and the information is transmitted to the critical band B.
  • information of the allowable noise level NL which is set only once, and information of the word lengths W1 to W4 corresponding to the number of allocated bits are transmitted. That is, in the subsequent decoding process, the information of the allowable noise level NL of the critical band B is Is transmitted, based on the information of the permissible noise level NL and the information of the code lengths W 1 to W 4 of the small blocks b 1 to b 4, respectively, for each of the small blocks b 1 to b 4.
  • the permissible noise level NL is determined for each critical band in consideration of human auditory characteristics, as described above.
  • the permissible noise level is substantially constant within one critical band. It is something that can be considered. Therefore, in each of the small blocks bl to b4 in the critical band B in FIG. 1A, the allowable noise level NL can be considered to be the same level.
  • the entire dynamic range is, for example, 120 dB and the floating coefficient is represented by the 6 bits
  • the floating coefficient has an accuracy of about 2 dB
  • the word length information is When represented by the above 4 bits, the code length information has an accuracy of about 6 dB. For this reason, in each of the small blocks b1 to b4 in FIG.
  • the tolerance noise level NL determined from the above has a deviation of about 2 dB steps as shown in FIG.
  • the allowable noise level NL usually falls within a range of about ⁇ 3 dB.
  • the allowable noise level NL is set in two steps: a rough coarse quantization common in the critical band, a fine level of each small block of the floating process in the critical band, and a quantization.
  • the permissible noise level NL expressed by quantization is set as a common value with high accuracy.
  • the allowable noise level NL is a 4-bit 1 og level
  • the allowable noise level NL that cannot be expressed by the 4-bit 10 g is finely expressed by the 2-bit 10 g. Therefore, in this embodiment, it is possible to divide about 6 dB into four parts to achieve 1.5 dB accuracy.
  • the number of bits can be reduced by selecting the high-precision parameter from the floating coefficient and the allowable noise level as shown in Will come. Table 2 shows how the number of bits is reduced in the example of FIG. 1A in comparison with Table 1 above.
  • the quantization of the information of the lengths W1 to W1 is not adaptive but uniform.
  • the block floating process is performed with a large block having a band wider than the critical band.
  • the block floating coefficient for each critical band and the information of the word length obtained for each critical band are required.
  • the block floating process is performed on a large block b having a band wider than the critical band as shown in FIG. 1B, the one large block b is used as information on the floating coefficient.
  • the allowable noise level NL can be obtained from the floating coefficient Fc1 and the word length W1
  • the remaining word length W2 is obtained from the allowable level NL and the floating coefficients Fc2 to Fc4.
  • ⁇ W4 will be able to know. For this reason, it is possible to adopt a high efficiency method that does not transmit the information of the remaining code lengths W2 to W4, and therefore, for the critical band B, the above three code lengths are used. The number of bits for transmitting the information of W2 to W4 can be reduced.
  • Table 3 shows how the number of bits is reduced in the example of FIG. 1B in comparison with Table 1 above.
  • a total of 22 bits, ie, 6 bits for c and 4 X 4 16 bits for code length W, will be transmitted. Therefore, when the number of transmission bits in the example in Table 2 is set to 100% and compared with the example in Table 3, in the example in Table 3, one band is 100% similarly, while 70% for 3 bands, 60% for 3 bands, 55% for 4 bands, etc. The lapse rate is improved. Therefore, it can be understood that the method of this embodiment is very effective.
  • step S4 when the block floating processing is performed in units of small blocks in a band narrower than the critical band, in the step S4, one step according to the number of bits allocated to each small block is performed.
  • step S5 and step S6 instead of the floating coefficient for each critical band, a predetermined level from the signal level range within the critical band is used in step S5 and step S6. (Based on the level based on the information), the information about the permissible noise level that is shifted to the lower side is calculated and transmitted. Specifically
  • the specified range of the allowable noise level is shifted for the following 5 reasons.
  • the allowable noise level is determined based on the masking amount in consideration of the human auditory characteristics, as described later, so that the allowable noise level is always a certain level with respect to the maximum 0 of the actual signal level. It will be low. For example, the permissible noise level may be taken about 26 dB lower than the signal level.
  • the signal level in the critical band is In this case, a quantization table is used which shifts the prescribed range toward the lower level by a predetermined level from the target range. This makes it possible to express the allowable noise level even with a small number of bits. In other words, even with a small number of bits, the allowable noise level can be expressed with the same resolution (accuracy) as when the above shift is not performed. Therefore, it is possible to reduce the number of bits for transmitting an allowable noise level.
  • the difference between the transmission noise level of the small block b1 given the above-mentioned code length W1 and the level closest to the above-mentioned allowable noise level NL is transferred together. . That is, the information indicating the difference from the level closest to the allowable noise level NL indicates which level within the range of approximately ⁇ 3 dB in FIG. 7 or the transmission noise level of each small block comes.
  • the information may be transmitted using, for example, the determination bit information of a 2-bit determination bit.
  • the transmission noise level of the small block b1 is obtained from the information of the floating coefficient Fc1 and the word length W1, and the difference between the transmission noise level and the level closest to the allowable noise level N (represented by the judgment bit) is obtained. Level difference), the level closest to the allowable noise level NL can be obtained.
  • the transmitted noise level of each of the small blocks bl to b4 is closest to the allowable noise level NL and does not deviate beyond the level of the noise level.
  • Table 4 shows how the number of bits is reduced in the example of FIG. 1 in comparison with Table 1 above, which makes it possible to obtain the word lengths W2 to W4. [Table 4]
  • the level that can be taken as the allowable noise level NL is smaller than the dynamic range. That is, since the minimum value of SZN is about 30 dB, the allowable noise level does not come from the peak value of the dynamic range (0 dB) to about 30 dB. Can be set to g.
  • FIG. 3 shows an example of a configuration to which the above-described allowable noise level is obtained, and further based on the allowable noise level, the signal is encoded with the above-mentioned number of allocated bits, to which the encoding method of the present embodiment is applied.
  • the digital audio data on the time axis supplied to the input terminal 1 is transmitted to the orthogonal transformation circuit 11.
  • the audio data on the time axis is converted into data on the frequency axis for each unit time (unit block), and a coefficient composed of a real component value Re and an imaginary component value Im is obtained. Data is obtained.
  • These coefficient data are transmitted to the amplitude / phase information generation circuit 12, and the amplitude / phase information generation circuit 12 obtains amplitude information Am and phase information Ph from the real component value Re and the imaginary component value Im. As a result, only the information of the amplitude information Am is output.
  • the above-mentioned allocated bit number information is obtained using only the amplitude information Am. I have to.
  • the amplitude information Am is first transmitted to the band division circuit 13.
  • the band dividing circuit 13 converts the input digital signal represented by the amplitude information Am into a so-called critical band. It is divided into widths (critical bands).
  • This critical bandwidth takes into account human auditory characteristics (frequency analysis capability). For example, the frequency band of 0 to 22 kHz is divided into 25 bands, and the higher the frequency band, the more the bandwidth is increased. It has been widely selected. That is, human hearing has characteristics like a kind of band pass filter, and the band divided by each filter is called a critical band.
  • the amplitude information Am for each band divided into the critical bands by the band division circuit 13 is transmitted to the sum detection circuit 14, respectively.
  • the energy for each band is the sum of the amplitude information Am in each band (peak or average of amplitude information Am or energy sum).
  • the output of the sum detection circuit 14, that is, the spectrum o of the sum of each band.
  • the bark spectrum SB of each band is as shown in FIG. 4, for example.
  • FIG. 4 which represent the number of bands of the critical bands 1 second band (B> ⁇ B 1 2).
  • the filter circuit 15 includes, for example, a plurality of delay elements for sequentially delaying input data, and a plurality of multipliers for multiplying the output from these delay elements by a filter coefficient (weighting function). 25 multipliers), and a sum adder for summing the outputs of the multipliers. The following multiplication is performed in each multiplier of the filter circuit 15.
  • a multiplier M corresponding to an arbitrary band has a filter coefficient of 1
  • a multiplier M-1 has a filter coefficient of 0.15
  • a multiplier M-2 has a filter coefficient of 0.019
  • M is any integer from 1 to 25.
  • the masking refers to a phenomenon in which a certain signal masks another signal and makes it inaudible due to human auditory characteristics.
  • This masking effect includes a masking effect for an audio signal on a time axis. And a masking effect for signals on the frequency axis. That is, even if noise is applied to the masked portion due to the masking effect, this noise will not be heard. For this reason, in the actual audio signal, the noise in the masked portion is regarded as an acceptable noise.
  • the subtracter 16 obtains a level corresponding to an allowable noise level described later in the convolved area.
  • the level corresponding to the permissible noise level is, as described later, a level at which the permissible noise level of each critical band is obtained by performing inverse convolution processing. It is.
  • the subtractor 16 is supplied with an allowance function (a function expressing a masking level) for obtaining the level H.
  • the level control is performed by increasing or decreasing the allowable function.
  • the permissible function is supplied from a function generating circuit 29 described later.
  • the level corresponding to the allowable noise level can be calculated by the following equation, where i is a number sequentially given from the lower band of the critical bandwidth.
  • n and a are constants, a> 0, and S is the intensity of the convolution processed bark vector, where (n ⁇ ai) is an allowable function.
  • the divider 17 is for decomposing the level ⁇ in the convolved region. Therefore, by performing the inverse composition processing, a masking spectrum can be obtained from the level. That is, this masking spectrum becomes an allowable noise spectrum. Note that the above inverse convolution process requires a complicated operation, but in this embodiment, the inverse convolution is performed using a simplified divider 17.
  • the masking spectrum is transmitted to the subtractor 19 via the combining circuit 18.
  • the output of the sum detection circuit 14, that is, the bark vector SB from the sum detection circuit 14 described above is supplied to the subtracter 19 via the delay circuit 21.
  • the bark spectrum SB is converted to the level of the masking spectrum MS. Below the indicated level will be masked.
  • the output of the subtracter 19 is sent to the ROM 30 via the allowable noise level correction circuit 20.
  • the ROM 30 stores a plurality of pieces of assigned bit number information used for quantization of the amplitude information Am.
  • the output of the subtraction circuit 19 (the energy of each band and the noise level setting means)
  • the number of bits allocated is output according to the level of the difference from the output. Therefore, in the quantization circuit 24, the amplitude information Am is quantized based on the allocated bit number information, and the output is output from the output terminal 2.
  • the delay circuit 21 delays the bark vector SB from the sum detection circuit 14 in consideration of the delay amount of each circuit before the synthesis circuit 18, and the delay circuit 23 It is provided to delay the amplitude information Am in consideration of the delay amount in each circuit before 0.
  • the minimum audible curve generation circuit 22 It is possible to combine the masking spectrum MS with the data indicating the human auditory characteristics as shown in FIG. At this minimum audible curve, if the absolute noise level is below this minimum audible curve, the noise will not be heard. Further, the minimum audible curve differs depending on, for example, the reproduction volume at the time of reproduction even if the coding is the same. However, in a realistic digital signal system, there is not much difference in the way music enters the 16-bit dynamic range, for example, so that the most audible sound around 4 kHz, If quantization noise is not heard, it is considered that quantization noise below the level of this minimum audible curve is not heard in other frequency bands.
  • the minimum audible curve RC and the masking vector MS are combined together.
  • the allowable noise level in this case can be set to a portion indicated by oblique lines in the drawing.
  • the 4 kHz level of the minimum audible curve is adjusted to the lowest level corresponding to, for example, 20 bits.
  • FIG. 6 also shows the signal spectrum SS.
  • the permissible noise level correction circuit 20 corrects the permissible noise level from the subtracter 19 based on information of a so-called equal loudness curve sent from the correction value determination circuit 28. That is, the correction value determination circuit 28 outputs correction value data for correcting the allowable noise level from the subtractor 19 based on information of a so-called equal loudness curve. The data is transmitted to the allowable noise level correction circuit 20. In this way, the allowable noise level from the subtractor 19 is corrected in consideration of the equal loudness curve.
  • the above equal loudness curve relates to human auditory characteristics.For example, the sound pressure of each frequency that sounds as loud as a pure tone of 1 kHz is obtained and connected by a curve. — 9—
  • the equal loudness curve is substantially the same as the minimum audible curve RC shown in FIG.
  • the sound pressure is 8 to 10 dB below 1 kHz and the same as 1 kHz.
  • the sound cannot be heard at the same level unless it is higher than the sound pressure at 1 kHz by about 15 dB. Therefore, it can be seen that noise exceeding the level of the minimum audible curve (allowable noise level) preferably has a frequency characteristic given by a curve corresponding to the equal loudness curve. From this, it can be understood that the correction of the allowable noise level in consideration of the equal loudness curve is suitable for human auditory characteristics.
  • a configuration in which the above-described minimum audible curve synthesizing process is not performed may be adopted. That is, in this case, the minimum audible curve generating circuit 22 and the synthesizing circuit 18 are not required, and the output from the subtracter 16 is immediately decomposed by the divider 17 and immediately thereafter. It will be transmitted to the subtractor 19.
  • the input digital signal is orthogonally transformed and divided into critical bands, and the signal component of each critical band is adaptively allocated based on the allowable noise level of each critical band.
  • the floating coefficients obtained by block-processing the signal components after the orthogonal transformation are transmitted.
  • Reduces the number of bits for the floating coefficient by transmitting the information on the allowable noise level for each critical band and the code length information of each small block instead of the floating coefficient for each critical band. If the block floating process is performed on a large block with a band wider than the critical band, By the this which is adapted to transmit information relating to floating coefficients every lock, it is possible to reduce the number of transmission bits.

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Theoretical Computer Science (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)
  • Signal Processing For Digital Recording And Reproducing (AREA)
  • Reduction Or Emphasis Of Bandwidth Of Signals (AREA)
  • Transmission Systems Not Characterized By The Medium Used For Transmission (AREA)
  • Measuring Pulse, Heart Rate, Blood Pressure Or Blood Flow (AREA)
  • Compression Or Coding Systems Of Tv Signals (AREA)

Abstract

L'invention concerne un appareil codant avec une grande efficacité des signaux numériques en convertissant des signaux d'entrée numériques en signaux sur un axe de fréquence, sous forme de blocs, et codant des signaux numériques d'entrée avec un nombre adéquat de bits d'affectation pour chacun desdits blocs, pour transmettre lesdits signaux. Les signaux numériques d'entrée sont soumis à la conversion orthogonale et sont divisés par une bande critique, codée avec un nombre adéquat de bits d'affectation, sur la base d'un niveau de bruit acceptable, et ils sont soumis au traitement flottant. Lorsque le traitement flottant doit être éxécuté par une pluralité de petits blocs d'une bande plus étroite que la bande critique, l'information de longueur de mot d'un petit bloc est émise à partir des petits blocs dans les bandes critiques, afin de réduire le nombre de bits pour la longueur de mot et pour comprimer les bits. En outre, lorsque le traitement flottant doit être éxécuté par de petits blocs de bandes plus étroites que les bandes critiques, le niveau de bruit acceptable et l'information de longueur de mot d'un petit bloc sont transmis pour chacune des bandes critiques, au lieu qu'il y ait une transmission du coefficient de flottement pour chacune des bandes critiques, cela afin de réduire le nombre de bits de transmission et de comprimer fortement les bits. De plus, lorsque le traitement flottant doit être éxécuté par de petits blocs de bandes plus étroites que les bandes critiques, l'information d'une longueur de mot correspondant au nombre de bits d'affectation est transmise, et l'information relative au niveau de bruit acceptable, dont la plage définie est déviée vers le bas d'une valeur prédéterminée à partir de la plage de niveau du signal à l'intérieur de la bande critique, est obtenue par l'intermédiaire d'une table de quantification, et est transmise au lieu qu'il y ait une transmission du coefficient de flottaison, celà afin de comprimer encore plus fortement les bits.
PCT/JP1992/000237 1991-03-29 1992-02-28 Procede de codage de signaux numeriques WO1992017942A1 (fr)

Priority Applications (8)

Application Number Priority Date Filing Date Title
RU92016428/09A RU2090973C1 (ru) 1991-03-29 1992-02-28 Способ кодирования сигналов
EP92906204A EP0531538B1 (fr) 1991-03-29 1992-02-28 Reduction de la taille d'information de formatage pour codage par sous-bandes
DK92906204T DK0531538T3 (da) 1991-03-29 1992-02-28 Fremgangsmåde til indkodning af digitale signaler
BR9204799A BR9204799A (pt) 1991-03-29 1992-02-28 Processo de codificacao para um sinal digital
DE69225100T DE69225100T2 (de) 1991-03-29 1992-02-28 Reduzierung der Zusaztinformation bei Teilbandkodierungsverfahren
AU13383/92A AU654533B2 (en) 1991-03-29 1992-02-28 Method of encoding digital signals
NO92924552A NO924552L (no) 1991-03-29 1992-11-25 Fremgangsmaate og anordning for koding av digitalt audiosignal
FI925400A FI109627B (fi) 1991-03-29 1992-11-27 Menetelmä digitaalisten signaalien koodausta varten

Applications Claiming Priority (4)

Application Number Priority Date Filing Date Title
JP3/91187 1991-03-29
JP3091188A JP3060577B2 (ja) 1991-03-29 1991-03-29 ディジタル信号符号化方法
JP3/91188 1991-03-29
JP3091187A JP3060576B2 (ja) 1991-03-29 1991-03-29 ディジタル信号符号化方法

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HU9203759D0 (en) 1993-04-28
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BR9204799A (pt) 1993-07-13
US5311561A (en) 1994-05-10
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DE69225100D1 (de) 1998-05-20
FI109627B (fi) 2002-09-13
ATE165198T1 (de) 1998-05-15
CN1032102C (zh) 1996-06-19
AU1338392A (en) 1992-11-02
AU654533B2 (en) 1994-11-10
HU213592B (en) 1997-08-28
EP0531538A1 (fr) 1993-03-17
FI925400A0 (fi) 1992-11-27
EP0531538B1 (fr) 1998-04-15
CN1065565A (zh) 1992-10-21
DE69225100T2 (de) 1998-08-06
RU2090973C1 (ru) 1997-09-20
KR100312664B1 (ko) 2002-12-26
EP0531538A4 (en) 1994-09-21
NO924552L (no) 1993-01-22
HUT63278A (en) 1993-07-28
CA2083709A1 (fr) 1992-09-30
PL294003A2 (en) 1992-11-30

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